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Autres articles (91)
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MediaSPIP v0.2
21 juin 2013, parMediaSPIP 0.2 est la première version de MediaSPIP stable.
Sa date de sortie officielle est le 21 juin 2013 et est annoncée ici.
Le fichier zip ici présent contient uniquement les sources de MediaSPIP en version standalone.
Comme pour la version précédente, il est nécessaire d’installer manuellement l’ensemble des dépendances logicielles sur le serveur.
Si vous souhaitez utiliser cette archive pour une installation en mode ferme, il vous faudra également procéder à d’autres modifications (...) -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
(Dés)Activation de fonctionnalités (plugins)
18 février 2011, parPour gérer l’ajout et la suppression de fonctionnalités supplémentaires (ou plugins), MediaSPIP utilise à partir de la version 0.2 SVP.
SVP permet l’activation facile de plugins depuis l’espace de configuration de MediaSPIP.
Pour y accéder, il suffit de se rendre dans l’espace de configuration puis de se rendre sur la page "Gestion des plugins".
MediaSPIP est fourni par défaut avec l’ensemble des plugins dits "compatibles", ils ont été testés et intégrés afin de fonctionner parfaitement avec chaque (...)
Sur d’autres sites (6680)
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ffmpeg audio convert not working
11 janvier 2012, par AndrewI cannot get the ffmpeg audio convert working on my site. The idea is that file should be converted when it's uploaded on the site.
I've got this on my upload form determining the audio file's format :
if(isset($_POST['audio']) && $_POST['audio'] != ''){
$ext1 = substr($_POST['audio'], -4);This is the best I've come up with for converting m4a to mp3 :
if(isset($_POST['audio']) && $_POST['audio'] != ''){ $file = $_POST['audio']; if($ext1==".m4a"){ $call="/usr/local/bin/ffmpeg -i ".$file." -ab 192k -f -acodec mp3";}
$convert = (popen("start /b ".$call, "r")); pclose($convert);The problem is, it won't convert. The path to ffmpeg is correct.
Now I may be way over my head with this one, but if there's a simple solution for this, I'd love to hear it.
EDIT.
With this :
if(isset($_POST['audio']) && $_POST['audio'] != ''){
$file = $_POST['audio'];
$ext1 = substr($_POST['audio'], -4); /*get the last 4 chars*/
$mp3 = echo 'mp3';
if($ext1=".m4a"){
"/usr/local/bin/ffmpeg -i \"". $file . "\" -ab 192k -y -f mp3 \"".ext1.$mp3."\"";
}
}I think I'm right on the money with conversion itself, but the form just loads infinitly when submitted. So I'm guessing the conversion is happening, but the form does not know when it's done. Any ideas on that ?
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FFMPEG 1.0 ANDROID NDK R8B
6 février 2013, par user1613977I have tried to compile the last version of ffmpeg with the android ndk r8b, but i always have errors and i dont know how to solve them. I am reading this tutorial http://www.roman10.net/how-to-build-ffmpeg-for-android/ but the script doesnt work with my environment, i am using this :
Ubuntu 12.04 LTS.
Intel Core i7.
Android ndk r8b
Last version ffmeg (1.0)Basically i want to encode films from one format to other.
And that is the script that i am using :
NDK=../android-ndk-r8b
PLATFORM=$NDK/platforms/android-8/arch-arm/
PREBUILT=$NDK/toolchains/arm-linux-androideabi-4.4.3/prebuilt/linux-x86
function build_r8b
{
bash configure \
--disable-shared \
--enable-static \
--enable-gpl \
--enable-version3 \
--enable-nonfree \
--disable-doc \
--disable-ffmpeg \
--disable-ffplay \
--disable-ffprobe \
--disable-ffserver \
--disable-avdevice \
--disable-avfilter \
--disable-postproc \
--enable-small \
--cross-prefix=$PREBUILT/bin/arm-linux-androideabi- \
--enable-cross-compile \
--target-os=linux \
--extra-cflags="-I$PLATFORM/usr/include" \
--extra-ldflags="-L$PLATFORM/usr/lib -nostdlib" \
--arch=arm \
--disable-symver \
--disable-debug \
--disable-stripping \
$ADDITIONAL_CONFIGURE_FLAG
sed -i 's/HAVE_LRINT 0/HAVE_LRINT 1/g' config.h
sed -i 's/HAVE_LRINTF 0/HAVE_LRINTF 1/g' config.h
sed -i 's/HAVE_ROUND 0/HAVE_ROUND 1/g' config.h
sed -i 's/HAVE_ROUNDF 0/HAVE_ROUNDF 1/g' config.h
sed -i 's/HAVE_TRUNC 0/HAVE_TRUNC 1/g' config.h
sed -i 's/HAVE_TRUNCF 0/HAVE_TRUNCF 1/g' config.h
make clean
make -j4 install
$PREBUILT/bin/arm-linux-androideabi-ar d libavcodec/libavcodec.a inverse.o
$PREBUILT/bin/arm-linux-androideabi-ld -rpath-link=$PLATFORM/usr/lib -L$PLATFORM/usr/lib -soname libffmpeg.so -shared -nostdlib -z,noexecstack -Bsymbolic --whole-archive --no-undefined -o $PREFIX/libffmpeg.so libavcodec/libavcodec.a libavformat/libavformat.a libavutil/libavutil.a libswscale/libswscale.a -lc -lm -lz -ldl -llog --warn-once --dynamic-linker=/system/bin/linker $PREBUILT/lib/gcc/arm-linux-androideabi/4.4.3/libgcc.a
}
#arm v7vfpv3
CPU=armv7-a
OPTIMIZE_CFLAGS="-mfloat-abi=softfp -mfpu=vfpv3-d16 -marm -march=$CPU "
PREFIX=./android/$CPU
ADDITIONAL_CONFIGURE_FLAG=
#build_one
build_r8bAnd that is the terminal result
jllarraz@ubuntu:~/Descargas/ffmpeg$ bash build_android.sh
/android-ndk-r8b/toolchains/arm-linux-androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi-gcc is unable to create an executable file.
C compiler test failed.
If you think configure made a mistake, make sure you are using the latest
version from Git. If the latest version fails, report the problem to the
ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net.
Include the log file "config.log" produced by configure as this will help
solving the problem.
make: execvp: ./version.sh: Permiso denegado
make: execvp: ./version.sh: Permiso denegado
HOSTCC doc/print_options.o
CC cmdutils.o
CC libavdevice/alldevices.o
CC libavdevice/avdevice.o
In file included from doc/print_options.c:104:0:
./libavcodec/options_table.h: En la función ‘show_codec_opts’:
./libavcodec/options_table.h:85:1: aviso: ‘sub_id’ es obsoleto (declarado en ./libavcodec/avcodec.h:1505) [-Wdeprecated-declarations]
./libavcodec/options_table.h:149:1: aviso: ‘luma_elim_threshold’ es obsoleto (declarado en ./libavcodec/avcodec.h:1740) [-Wdeprecated-declarations]
./libavcodec/options_table.h:150:1: aviso: ‘chroma_elim_threshold’ es obsoleto (declarado en ./libavcodec/avcodec.h:1747) [-Wdeprecated-declarations]
./libavcodec/options_table.h:284:1: aviso: ‘color_table_id’ es obsoleto (declarado en ./libavcodec/avcodec.h:1986) [-Wdeprecated-declarations]
./libavcodec/options_table.h:307:1: aviso: ‘inter_threshold’ es obsoleto (declarado en ./libavcodec/avcodec.h:2049) [-Wdeprecated-declarations]
./libavcodec/options_table.h:312:1: aviso: ‘quantizer_noise_shaping’ es obsoleto (declarado en ./libavcodec/avcodec.h:2056) [-Wdeprecated-declarations]
cmdutils.c:52:21: error fatal: version.h: No existe el archivo o el directorio
compilación terminada.
make: *** [cmdutils.o] Error 1
make: *** Se espera a que terminen otras tareas....
build_android.sh: línea 59: ../android-ndk-r8b/toolchains/arm-linux-androideabi- 4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi-ar: Permiso denegado
build_android.sh: línea 60: ../android-ndk-r8b/toolchains/arm-linux-androideabi-4.4.3/prebuilt/linux-x86/bin/arm-linux-androideabi-ld: Permiso denegado -
Convert stream to a stream on the fly with ffmpeg/avconv
6 juillet 2014, par ApacciI’m trying to convert flv stream to another mp3 or ogg stream. The point is I want to use mp3/ogg stream on HTML5 page, because flv stream can’t be played with HTML5. So, I configured avserver.conf :
Port 8090
BindAddress 0.0.0.0
MaxHTTPConnections 2000
MaxClients 1000
MaxBandwidth 1000
CustomLog -
<feed>
File /tmp/feed1.ffm
FileMaxSize 200K
ACL allow 127.0.0.1
</feed>
<stream>
Feed feed1.ffm
Format ogg
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</stream>
<stream>
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</stream>
<redirect>
URL http://www.libav.org/
</redirect>the command I use :
avconv -i http://some_stream_service?format=flv -acodec mp3 http://localhost:8090/feed1.ffmconsole output :
avconv version 9.13-6:9.13-0ubuntu0.14.04.1+fdkaac, Copyright (c) 2000-2014 the Libav developers
built on May 10 2014 17:26:31 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
[flv @ 0x126e020] max_analyze_duration reached
Input #0, flv, from 'http://some_stream_service?format=flv':
Metadata:
encoder : Lavf52.87.1
Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
Stream #0.0: Audio: aac, 44100 Hz, stereo, fltp, 31 kb/s
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
Metadata:
encoder : Lavf54.20.4
Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, fltp
Stream mapping:
Stream #0:0 -> #0:0 (aac -> libmp3lame)
Press ctrl-c to stop encoding
size= 1080kB time=66.14 bitrate= 133.8kbits/sIt looks like working, but I keep getting 0 bytes on web page
http://localhost:8090/test1.ogg. Does anybody have ideas what I’m doing wrong ?
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